Asterisk builds two channels and bridges them together. If the codecs mis-match then it must transcode, the negotiation on the Zap end is done seperately from the SIP end, so it does not care what your handset decided on.
If you want ulaw, use ulaw, not g729 (on any call leg). You won't be able to mix and match codecs between calls, choose one for all calls and stick with it. -----Original Message----- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jeremy Mann Sent: Tuesday, April 15, 2008 08:24 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Zap Codec I guess that's my frustration, I don't want it g729, I want it ulaw, I just wish Zap did codec negotiation from the client. It'd be a nice option instead of automatically trying to translate if it's not ulaw. Could save some processor overhead(obviously at the expense of bandwidth). -----Original Message----- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric Wieling Sent: Tuesday, April 15, 2008 10:15 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Zap Codec If you want to get a G729 call to go via Zap you must purchase a G729 license. No amount of discussion is going to change that. Jeremy Mann wrote: > Sadly you are correct: > > > -- Executing [EMAIL PROTECTED]:4] Set("SIP/156-083514c0", "_SIP_CODEC=ulaw") in new stack > -- Executing [EMAIL PROTECTED]:5] NoOp("SIP/156-083514c0", "4") in new stack > -- Executing [EMAIL PROTECTED]:6] NoOp("SIP/156-083514c0", """) in new stack > -- Executing [EMAIL PROTECTED]:7] Dial("SIP/156-083514c0", "") in new stack > [Apr 15 09:51:25] WARNING[17434]: channel.c:3265 ast_request: No translator path exists for channel type Zap (native 76) to 256 > [Apr 15 09:51:25] WARNING[17434]: app_dial.c:1191 dial_exec_full: Unable to create channel of type 'Zap' (cause 58 - Bearer capability not available) > == Everyone is busy/congested at this time (1:0/0/1) > -- Executing [EMAIL PROTECTED]:8] Hangup("SIP/156-083514c0", "") in new stack > > -----Original Message----- > From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric Wieling > Sent: Tuesday, April 15, 2008 9:43 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] Zap Codec > > That would work just spiffy if you are calling another SIP device, but > by the time the call gets to that point in the dialplan the codec of the > originating device has already been chosen and set in stone. > > Tilghman Lesher wrote: >> On Tuesday 15 April 2008 07:40:47 Jeremy Mann wrote: >>> But I want my polycom to attempt g729 on SIPPEER-SIPPEER calls, I only want >>> ulaw used when SIPPEER-ZAP is the case. >> Set(_SIP_CODEC=ulaw) >> Dial(Zap/g0/...) >> > > -- > Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS, > T-1, PRI, Frame Relay, Linux, and network design. Based near > Birmingham, AL. Now accepting clients worldwide. > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > This e-mail, facsimile, or letter and any files or attachments transmitted with it contains information that is confidential and privileged. This information is intended only for the use of the individual(s) and entity(ies) to whom it is addressed. If you are the intended recipient, further disclosures are prohibited without proper authorization. If you are not the intended recipient, any disclosure, copying, printing, or use of this information is strictly prohibited and possibly a violation of federal or state law and regulations. If you have received this information in error, please notify Texas Health Management Group immediately at 1-817-310-4999. Texas Health Management Group, its subsidiaries, and affiliates hereby claim all applicable privileges related to this information. > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > -- Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS, T-1, PRI, Frame Relay, Linux, and network design. Based near Birmingham, AL. Now accepting clients worldwide. _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This e-mail, facsimile, or letter and any files or attachments transmitted with it contains information that is confidential and privileged. This information is intended only for the use of the individual(s) and entity(ies) to whom it is addressed. If you are the intended recipient, further disclosures are prohibited without proper authorization. If you are not the intended recipient, any disclosure, copying, printing, or use of this information is strictly prohibited and possibly a violation of federal or state law and regulations. If you have received this information in error, please notify Texas Health Management Group immediately at 1-817-310-4999. Texas Health Management Group, its subsidiaries, and affiliates hereby claim all applicable privileges related to this information. _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users